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Joshua Billings

2009-Jun-30 22:00 UTC

[asterisk-users] Authentication Issue Between Servers

I've got an issue where I am trying to route calls between Asterisk 
Servers.  I can route calls inbound to a server but seem to have an 
authentication issue going out over the same sip account.  It appears 
that my server isn't sending the second invite after proxy 
authentication request.  I can't figure out why; any ideas would be 
greatly appreciated.  Thanks!

- Josh


Here is my sip.conf:

[general]
context = default
allowoverlap = no
bindport = 5060
bindaddr = 0.0.0.0
srvlookup = yes
externip = 172.21.235.2
localnet = 172.21.235.2/255.255.0.0
dtmfmode = rfc2833
relaxdtmf = yes
disallow = all
allow = ulaw
allow = gsm
maxexpirey = 30
defaultexpirey = 180
relaxdtmf=yes
canreinvite = no
nat = 0
UserAgent = Asterisk
echocancel = yes
echocancelwhenbridge = yes
t38pt_udptl = no

[trunk]
type = friend
callwaiting = yes
caller id contact context = default
fullname group hasagent = no
hasdirectory = yes
hasiax = no
hasmanager = no
hassip = yes
host = 172.21.235.1
secret = [password]
threewaycalling = yes
registersip = yes
canreinvite = no
nat = no
dtmfmode = rfc2833
registeriax = no
disallow = all
allow = gsm
register=>trunk:[password]@172.21.235.1


Here is the applicable portion of extensions.conf:

[default]
exten = _5XX,1,Dial(SIP/trunk/${EXTEN},,Tt)


Here is the SIP Debug output:

INVITE sip:510 at 172.21.235.1 SIP/2.0
Via: SIP/2.0/UDP 172.21.235.2:5060;branch=z9hG4bK78d4e8d7;rport
From: "Marci" <sip:3874 at 172.21.235.2>;tag=as5951033c
To: <sip:510 at 172.21.235.1>
Contact: <sip:3874 at 172.21.235.2>
Call-ID: 430c49156ce4a7500b1fa57807b5acf1 at 172.21.235.2
CSeq: 102 INVITE
User-Agent: Asterisk
Max-Forwards: 70
Date: Tue, 30 Jun 2009 19:09:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 239

v=0
o=root 11411 11411 IN IP4 172.21.235.2
s=session
c=IN IP4 172.21.235.2
t=0 0
m=audio 11486 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
^@
^[[KWBPBXFG000304*CLI>
<--- SIP read from 172.21.235.1:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 
172.21.235.2:5060;branch=z9hG4bK78d4e8d7;received=172.21.235.2;rport=5060
From: "Marci" <sip:3874 at 172.21.235.2>;tag=as5951033c
To: <sip:510 at 172.21.235.1>;tag=as045cd609
Call-ID: 430c49156ce4a7500b1fa57807b5acf1 at 172.21.235.2
CSeq: 102 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="4c4374da"
Content-Length: 0


<------------->
^@
^[[KWBPBXFG000304*CLI>
--- (11 headers 0 lines) ---
^@
^[[KWBPBXFG000304*CLI>
Transmitting (NAT) to 172.21.235.1:5060:
ACK sip:510 at 172.21.235.1 SIP/2.0
Via: SIP/2.0/UDP 172.21.235.2:5060;branch=z9hG4bK78d4e8d7;rport
From: "Marci" <sip:3874 at 172.21.235.2>;tag=as5951033c
To: <sip:510 at 172.21.235.1>;tag=as045cd609
Contact: <sip:3874 at 172.21.235.2>
Call-ID: 430c49156ce4a7500b1fa57807b5acf1 at 172.21.235.2
CSeq: 102 ACK
User-Agent: Asterisk
Max-Forwards: 70
Content-Length: 0

---
^@
^[[KWBPBXFG000304*CLI>
[Jun 30 14:09:25] NOTICE[11434]: chan_sip.c:12253 
handle_response_invite: ^@Failed to authenticate on INVITE to
'"Marci"
<sip:3874 at 172.21.235.2>;tag=as5951033c'
^@
^[[KWBPBXFG000304*CLI>
Really destroying SIP dialog 
'430c49156ce4a7500b1fa57807b5acf1 at 172.21.235.2' Method: INVITE
^@
^[[KWBPBXFG000304*CLI>
Really destroying SIP dialog 
'0fe5f50f7674160d2ab3522f09060d46 at 127.0.0.1' Method: REGISTER
^@
^[[KWBPBXFG000304*CLI>

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Joshua Billings

2009-Jul-02 15:37 UTC

head link

[asterisk-users] Authentication Issue Between Servers

No one ever responded to this inquiry but I figured out what the issue 
was.  I thought I would respond with the solution just in case someone 
runs into the same issue in the future.

Firstly, when setting up trunking between servers the "username ="
field
is not optional. :)  Also, I had a lot of extra fields in place that I 
didn't need but hadn't taken the time to remove.  I have developed the 
opinion that config files should be kept as lean as possible.  Here is 
the revised SIP peer configuration from sip.conf:

[trunk]
type = friend
username = trunk
callerid context = default
host = 172.21.235.1
secret = password
canreinvite = no
disallow = all
allow = gsm



Joshua Billings wrote:> I've got an issue where I am trying to route calls between Asterisk 
> Servers.  I can route calls inbound to a server but seem to have an 
> authentication issue going out over the same sip account.  It appears 
> that my server isn't sending the second invite after proxy 
> authentication request.  I can't figure out why; any ideas would be 
> greatly appreciated.  Thanks!
>
> - Josh
>
>
> Here is my sip.conf:
>
> [general]
> context = default
> allowoverlap = no
> bindport = 5060
> bindaddr = 0.0.0.0
> srvlookup = yes
> externip = 172.21.235.2
> localnet = 172.21.235.2/255.255.0.0
> dtmfmode = rfc2833
> relaxdtmf = yes
> disallow = all
> allow = ulaw
> allow = gsm
> maxexpirey = 30
> defaultexpirey = 180
> relaxdtmf=yes
> canreinvite = no
> nat = 0
> UserAgent = Asterisk
> echocancel = yes
> echocancelwhenbridge = yes
> t38pt_udptl = no
>
> [trunk]
> type = friend
> callwaiting = yes
> caller id > contact > context = default
> fullname > group > hasagent = no
> hasdirectory = yes
> hasiax = no
> hasmanager = no
> hassip = yes
> host = 172.21.235.1
> secret = [password]
> threewaycalling = yes
> registersip = yes
> canreinvite = no
> nat = no
> dtmfmode = rfc2833
> registeriax = no
> disallow = all
> allow = gsm
> register=>trunk:[password]@172.21.235.1
>
>
> Here is the applicable portion of extensions.conf:
>
> [default]
> exten = _5XX,1,Dial(SIP/trunk/${EXTEN},,Tt)
>
>
> Here is the SIP Debug output:
>
> INVITE sip:510 at 172.21.235.1 SIP/2.0
> Via: SIP/2.0/UDP 172.21.235.2:5060;branch=z9hG4bK78d4e8d7;rport
> From: "Marci" <sip:3874 at 172.21.235.2>;tag=as5951033c
> To: <sip:510 at 172.21.235.1>
> Contact: <sip:3874 at 172.21.235.2>
> Call-ID: 430c49156ce4a7500b1fa57807b5acf1 at 172.21.235.2
> CSeq: 102 INVITE
> User-Agent: Asterisk
> Max-Forwards: 70
> Date: Tue, 30 Jun 2009 19:09:25 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Type: application/sdp
> Content-Length: 239
>
> v=0
> o=root 11411 11411 IN IP4 172.21.235.2
> s=session
> c=IN IP4 172.21.235.2
> t=0 0
> m=audio 11486 RTP/AVP 3 101
> a=rtpmap:3 GSM/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
>
> ---
> ^@
> ^[[KWBPBXFG000304*CLI>
> <--- SIP read from 172.21.235.1:5060 --->
> SIP/2.0 407 Proxy Authentication Required
> Via: SIP/2.0/UDP 
> 172.21.235.2:5060;branch=z9hG4bK78d4e8d7;received=172.21.235.2;rport=5060
> From: "Marci" <sip:3874 at 172.21.235.2>;tag=as5951033c
> To: <sip:510 at 172.21.235.1>;tag=as045cd609
> Call-ID: 430c49156ce4a7500b1fa57807b5acf1 at 172.21.235.2
> CSeq: 102 INVITE
> User-Agent: Asterisk
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", 
> nonce="4c4374da"
> Content-Length: 0
>
>
> <------------->
> ^@
> ^[[KWBPBXFG000304*CLI>
> --- (11 headers 0 lines) ---
> ^@
> ^[[KWBPBXFG000304*CLI>
> Transmitting (NAT) to 172.21.235.1:5060:
> ACK sip:510 at 172.21.235.1 SIP/2.0
> Via: SIP/2.0/UDP 172.21.235.2:5060;branch=z9hG4bK78d4e8d7;rport
> From: "Marci" <sip:3874 at 172.21.235.2>;tag=as5951033c
> To: <sip:510 at 172.21.235.1>;tag=as045cd609
> Contact: <sip:3874 at 172.21.235.2>
> Call-ID: 430c49156ce4a7500b1fa57807b5acf1 at 172.21.235.2
> CSeq: 102 ACK
> User-Agent: Asterisk
> Max-Forwards: 70
> Content-Length: 0
>
> ---
> ^@
> ^[[KWBPBXFG000304*CLI>
> [Jun 30 14:09:25] NOTICE[11434]: chan_sip.c:12253 
> handle_response_invite: ^@Failed to authenticate on INVITE to
'"Marci"
> <sip:3874 at 172.21.235.2>;tag=as5951033c'
> ^@
> ^[[KWBPBXFG000304*CLI>
> Really destroying SIP dialog 
> '430c49156ce4a7500b1fa57807b5acf1 at 172.21.235.2' Method: INVITE
> ^@
> ^[[KWBPBXFG000304*CLI>
> Really destroying SIP dialog 
> '0fe5f50f7674160d2ab3522f09060d46 at 127.0.0.1' Method: REGISTER
> ^@
> ^[[KWBPBXFG000304*CLI>
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